Bourvine,
>> So, why won't we save the big bucks we pay them, hire two professionals
> (who cost less) and support an open source code by ourselves? This way
> we depend on ourselves only.
>>>> Thanks, __Yehavi:
I remember hearing University of Pennsylvania have been using Asterisk
for sometime. I am not certain where I came across that information,
but google confirmed it as a fact. And you may need to ask for more
details from Digium as they worked together, or call the school. I am
relatively certain they would share their experience. The deployment
was of 15,000 extensions, just about what you have in mind. Below is
some articles.
http://www.networkworld.com/news/2007/071707-open-source-voip.htmlhttp://www.digium.com/en/company/casestudies/viewcasestudies/University-of-Pennsylvania
William
>>>>> 2008/11/21 Grygoriy Dobrovolskyy <megahohol at gmail.com>
>>>> 2008/11/21 Yehavi Bourvine <yehavi.bourvine at gmail.com>
>> Hello,
>>>> Our university has to upgrade soon its old Nortel PBX's which
> holds around 10,000 extensions tied to 5 PBXes. Up to now we thought
> about commercial solutions but now there is a window openning for open
> source solution. However, I need examples to convince that this solution
> is feasible, and preferably at other universities.
>>>> Are there any pointers for such installations?
>>>> Thanks! __Yehavi:
>>>> _______________________________________________
> -- Bandwidth and Colocation Provided by
>http://www.api-digital.com <http://www.api-digital.com/> --
>> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users>>> Hello very interesting project you have, however asterisk is not
> a registry server, i suggest that you use opensips/opense/kamalio for
> your registrar, from where you dispatch to you asterisk servers, inside
> a good environment with a controlled network and nice tagged voip flow
> you could acheve a good results.
>>> _______________________________________________
> -- Bandwidth and Colocation Provided by
>http://www.api-digital.com <http://www.api-digital.com/> --
>> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users>>>>>>> -----------------------------------------
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>> Message: 9
> Date: Fri, 21 Nov 2008 09:46:13 -0500
> From: Alex Balashov <abalashov at evaristesys.com>
> Subject: Re: [asterisk-users] Large Asterisk installations (~10, 000
> extensions), preferably at universities
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4926C9B5.8080902 at evaristesys.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>> Jason Aarons (US) wrote:
>>> Just switching from Nortel to something else may not eliminate
>> hardware/software failures, or prevent those without experience from
>> pushing the enter key at the wrong time.
>> One also has to keep in mind - Asterisk, like any large open-source
> project, gets a lot more QA, patches and bug fixes than any commercial
> product sold in the intra-industrial channel (i.e. excluding consumer
> mass-market stuff) ever will! It has a massive installed base, many
> users reporting bugs through an open and easy to understand process, and
> a large community either directly or derivatively involved in
> contributing fixes and testing code.
>> How much installed base from which to harness that kind of large-scale
> technical feedback does Nortel have? Avaya? Cisco?
>> Asterisk has by far the best QA mechanism. In terms of potential bugs
> that impact "mission-critical" availability, I would feel better using
> it than any of these black-box, proprietary vendor solutions any day.
>> --
> Alex Balashov
> Evariste Systems
> Web : http://www.evaristesys.com/> Tel : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>>>> ------------------------------
>> Message: 10
> Date: Fri, 21 Nov 2008 15:46:59 +0100
> From: Philipp Kempgen <philipp.kempgen at amooma.de>
> Subject: Re: [asterisk-users] A way to run extenrnotify when IMAP
> events take place...
> To: Asterisk Users <asterisk-users at lists.digium.com>
> Message-ID: <4926C9E3.4070707 at amooma.de>
> Content-Type: text/plain; charset=ISO-8859-1
>> Danny Nicholas schrieb:
>> Here is a "Dirty" solution - create a PERL or other script to "listen" for
>> changes to voicemail DB/Dir. When VM is deleted, launch script to turn off
>> Cisco MWI (should be simple since you are turning on with script). Not
>> "Best" solution, just workable one.
>> Yeah. If all else should fail there are various dirty solutions
> such as listening to events on the manager interface, INotify,
> implementing a SMDI interface yourself ...
>> Philipp Kempgen
>> --
>http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com> Amooma GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de> Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> --
>>>> ------------------------------
>> Message: 11
> Date: Fri, 21 Nov 2008 09:47:57 -0500
> From: Alex Balashov <abalashov at evaristesys.com>
> Subject: Re: [asterisk-users] Large Asterisk installations (~10, 000
> extensions), preferably at universities
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4926CA1D.4070902 at evaristesys.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>> Alex Balashov wrote:
>> Jason Aarons (US) wrote:
>>>>> Just switching from Nortel to something else may not eliminate
>>> hardware/software failures, or prevent those without experience from
>>> pushing the enter key at the wrong time.
>>>> One also has to keep in mind - Asterisk, like any large open-source
>> project, gets a lot more QA, patches and bug fixes than any commercial
>> product sold in the intra-industrial channel (i.e. excluding consumer
>> mass-market stuff) ever will! It has a massive installed base, many
>> users reporting bugs through an open and easy to understand process, and
>> a large community either directly or derivatively involved in
>> contributing fixes and testing code.
>>>> How much installed base from which to harness that kind of large-scale
>> technical feedback does Nortel have? Avaya? Cisco?
>>>> Asterisk has by far the best QA mechanism. In terms of potential bugs
>> that impact "mission-critical" availability, I would feel better using
>> it than any of these black-box, proprietary vendor solutions any day.
>>>> Also, if there is a show-stopping bug, it can be addressed in a
> relatively expedient manner, especially if you are paying Digium for
> support.
>> With the other guys, you're going to have to wait for Service Pack 8
> Patchlevel 4 Release 2 Build 3789 in 12-24 months. It might have a fix.
> Maybe.
>> --
> Alex Balashov
> Evariste Systems
> Web : http://www.evaristesys.com/> Tel : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>>>> ------------------------------
>> Message: 12
> Date: Fri, 21 Nov 2008 09:53:36 -0500
> From: Alex Balashov <abalashov at evaristesys.com>
> Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000
> extensions), preferably at universities
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4926CB70.3040701 at evaristesys.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>> Al Baker wrote:
>>> Remember - You are going from a CARRIER GRADE purpose built piece of
>> hardware with Software built under a rigid CMM with extensive
>> "soak-testing" to software that has been developed under , shall we say,
>> a somewhat less rigid and stringent methodology.
>> You will be moving from an environment supported by hundreds of highly
>> trained people, some with decades of TELCO experience
>> to one where you support comes from a somewhat less seasoned group of
>> individuals.
>> But in choosing "carrier grade" (everyone calls their stuff that)
> vendors you are also going to a much smaller installed base and much
> lower total reporting and QA pool. I would take the sheer number and
> dynamism of the Asterisk installed base over their comparatively limited
> deployments, even if we grant the unsubstantiated premise that the
> latter is developed under a less rigid and stringent methodology.
>> Let me put it this way: if I wrote a piece of software and sold it to
> 10 customers, it won't matter for overall product quality that I fix the
> problems they report and maintain it for them under the guidance of a
> "rigid" and "stringent" methodology. That's nice. Hope it fixes their
> problems. It is really of comparatively minor benefit to prospective
> future adopters. It's not nearly as valuable as simply doing the best I
> can with bug reports and test cases from hundreds of users.
>> --
> Alex Balashov
> Evariste Systems
> Web : http://www.evaristesys.com/> Tel : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>>>> ------------------------------
>> Message: 13
> Date: Fri, 21 Nov 2008 11:59:29 -0300
> From: "Sebastian Milioto" <smilioto at gmail.com>
> Subject: [asterisk-users] Ping
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <e6e7910f0811210659m7dc9d8b7t4c171a9093b59c95 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>> Ping
> -------------- next part --------------
> An HTML attachment was scrubbed...
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>> Message: 14
> Date: Fri, 21 Nov 2008 10:04:57 -0500
> From: "Noah Miller" <noahisaacmiller at gmail.com>
> Subject: Re: [asterisk-users] Limit the number of users in a meetme
> conference?
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <8699dcab0811210704w2ec131eepdf7fc0ae18c10e42 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>> Hi Dan -
>>>> I found the "maxusers" defined in meetme.c, but I'm
>>> not sure how this value is set. Does anybody know
>>> if one can limit the number of users permitted in a
>>> meetme conference? I know there's MeetmeCount(), but
>>> I'd rather avoid the dialplan logic and just set
>>> maxusers instead.
>>>> That feature is primarily used with RealTime conferences.
>> The maxusers value is read from a database and enforced
>> on RealTime enable conferences. This presumes you are
>> looking at 1.6.X or Trunk code...
>> Ah. No realtime for me, so I guess I'll just stick with using
> MeetmeCount() in the dialplan. Thanks for the info!
>>> - Noah
>>>> ------------------------------
>> Message: 15
> Date: Fri, 21 Nov 2008 10:29:02 -0500
> From: "Noah Miller" <noahisaacmiller at gmail.com>
> Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000
> extensions), preferably at universities
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <8699dcab0811210729i29e38cbcjd4c6542a02fc983e at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>>> Due diligence is required on anything 10,000 people are going to be
>> pounding on. Undersizing is common,
>> I think due diligence is THE key with any open source solution,
> including asterisk. I'll admit that I pretty badly screwed up one
> asterisk installation because I didn't adequately prepare it (shipped
> it to the customer and had their IT staff install - bad plan). And
> while I've never done a system anywhere near 10K extensions, I've had
> good experiences with some large-ish installations because I budgeted
> in the time for research and testing.
>> I know that in the past there have been people on this list who have
> done very large scale asterisk deployments. Not sure if any of them
> are still around to comment.
>> With that many extensions, I'll second using a SIP registrar like
> Freeswitch or OpenSer. Just use asterisk to provide the services.
>>>> and is only one of the roads that
>> leads to Hell (I prefer Patterson Lake Road myself since I drive in from
>> the North East).
>> Hmm. You must live near Ann Arbor.
>>> - Noah
>>>> ------------------------------
>> Message: 16
> Date: Fri, 21 Nov 2008 10:32:51 -0500
> From: Alex Balashov <abalashov at evaristesys.com>
> Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000
> extensions), preferably at universities
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4926D4A3.7000306 at evaristesys.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>> Noah Miller wrote:
>>> With that many extensions, I'll second using a SIP registrar like
>> Freeswitch or OpenSer. Just use asterisk to provide the services.
>> Third.
>>> --
> Alex Balashov
> Evariste Systems
> Web : http://www.evaristesys.com/> Tel : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>>>> ------------------------------
>> Message: 17
> Date: Fri, 21 Nov 2008 07:36:27 -0800
> From: "Roderick A. Anderson" <raanders at acm.org>
> Subject: [asterisk-users] [SOLVED] TDM400 (?) zap hangup
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4926D57B.5050501 at acm.org>
> Content-Type: text/plain; charset=UTF-8; format=flowed
>> Roderick A. Anderson wrote:
>> And if that ain't confusing I don't know what would be.
>>>> I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago
>> and ended up never using it. Passed it along to a friend who is having
>> some problems with it. (He isn't on this list.)
>>>> We've both tried searches using Google but haven't been able to find
>> anything that helps. So this is more a question of
>> what-the-heck-should-we-be-searching-for. :-)
>>>> The TDM400 works taking inbound calls and gives a dial tone when the
>> phone is picked up but as soon as a key is pressed the line (Asterisk
>> says) hangs up. Asterisk is configured based on a working system but
>> that one only has one module inbound (FXO?) The outbound settings are
>> based on docs from voip-info.org.
>>>> Does this ring a bell for anyone? No pun intended.
>>>> Unfortunately the system is 35 miles away and I haven't got the logs
>> handy so I can't provide more right now. Just hoping for a clue on
>> search terms.
>> Thanks to Tzafrir Cohen and Jared Smith we've solved the problem.
>> It was a "A Series of Unfortunate Events". The main one was, there was
> no (and then an incorrect) context= for the ZAP channel. The incorrect
> one came about because of a miss-communication while testing. We were
> able to dial-out but the logic in the dialplan to select a context for
> local calls, toll-free, etc. was missing. Once we got the channel set
> to the correct context all was well.
>>> Again thanks,
> Rod
> --
>> TIA,
>> Rod
>>>>> ------------------------------
>> Message: 18
> Date: Fri, 21 Nov 2008 10:42:12 -0500
> From: "Matt Florell" <astmattf at gmail.com>
> Subject: Re: [asterisk-users] How long will Asterisk 1.4.x
> supported/maintained
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <61575c810811210742j6080a6d8q8018aa202d02d687 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>> On 11/20/08, Steve Totaro <stotaro at totarotechnologies.com> wrote:
>> On Thu, Nov 20, 2008 at 3:38 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
>> > On Thu, Nov 20, 2008 at 08:25:54AM +0100, Olivier wrote:
>> >> 2008/11/17 Philipp Kempgen <philipp.kempgen at amooma.de>
>> >>
>> >> > Tilghman Lesher schrieb:
>> >> > > On Thursday 13 November 2008 08:16:42 Klaus Darilion wrote:
>> >> > >> Is there somewhere a statement from Digium how long they will support
>> >> > >> Asterisk 1.4?
>> >> > >
>>>> 0>> > > There is no statement, because we haven't even discussed when
>>>> the EOL for
>> >> > > 1.4 will be reached. Certainly that means it won't happen for at least
>> >> > the
>> >> > > next 60 days, but beyond that, I really don't know.
>> >> >
>> >> > For the average non-techie user who does not want to compile
>> >> > themselves that may sound funny (if not scary).
>> >> >
>> >> > When Debian Lenny (featuring Asterisk 1.4) is finally going to be
>> >> > released that version might not even be supported any more.
>> >>
>> >>
>> >> I think to a large extend, Asterisk is not to be considered as binary
>> >> distributed at all, as many hardware it supports is not directly managed by
>> >> kernel team.
>> >
>> > Interesting consideration. Debian Etch and RHEL5 are based on kernel
>> > 2.6.18, but support quite a few hardware devices not included in that
>> > kernel.
>> >
>> > If this issue bothers you, please help test the alternative timing
>> > mechanism support now included in trunk.
>> >
>> > --
>> > Tzafrir Cohen
>> > icq#16849755 jabber:tzafrir.cohen at xorcom.com>> > +972-50-7952406 mailto:tzafrir.cohen at xorcom.com>> > http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
>> >
>>>>>> I still compile and install 1.2 for the most part, for call centers
>> and large systems.
>>>> The few 1.4 installs that I have done have been for "medium" sized
>> PBXs, say 50-70 phones/users and they have been trouble free for the
>> most part. Safe_asterisk may make some troubles transparent.
>>>> I am not really sure what 1.4 has over 1.2 for the average PBX installation.
>>>> Then you have the OpenPBX guys who forked 1.2, I know they have added
>> functionality to 1.2, but the following puts me off. Perhaps
>> vaporware, perhaps not, it all relies on the devs. You also have
>> people like Matt Florell who have continued to add functionality to
>> 1.2 but since Digium won't take them, or the dev doesn't want to sign
>> over their first born, they are hard to come by but certainly out
>> there.
>>>> 1.4 may follow the same path, being forked.
>>>> 1.6 is not on my radar.
>>>>>> --
>> Thanks,
>> Steve Totaro
>> +18887771888 (Toll Free)
>> +12409381212 (Cell)
>> +12024369784 (Skype)
>> Hello,
>> We really just maintain a set of patches for 1.2 (just updated
> waitforsilence a couple weeks ago in fact) and we regularly install
> 1.2.30.2 in call center setups. It is rock solid and extremely proven
> in high-call-volume situations.
>> We have started installing 1.4.21.2 on some systems that are not high
> load as well (1.4.22 has some strange issues with it we have noticed)
> because we do have clients requesting to use 1.4 for some of the nicer
> PBX functionality that it has as well as better SIP support.
>> We test 1.6 periodically and we are very much looking forward to some
> of the great new features of it, but it crashes very quickly when
> trying to use it in call center situations. just keep in mind that in
> my opinion the 1.4 tree did not become usable until 1.4.18 when most
> of the major bugs were finally fixed.
>> MATT---
>>>> ------------------------------
>> Message: 19
> Date: Fri, 21 Nov 2008 17:42:17 +0200
> From: "Atis Lezdins" <atis at iq-labs.net>
> Subject: Re: [asterisk-users] Ping
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <670f60170811210742v4d9baf35pc58f7f5db5cd3d09 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>> On Fri, Nov 21, 2008 at 4:59 PM, Sebastian Milioto <smilioto at gmail.com> wrote:
>> Ping
>>>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users>>>> Pong
>> GMail's preview looks fun - "Ping -- Bandwidth and Colocation Provided
> by http://www.api-digital.com">> Regards,
> Atis
>>> --
> Atis Lezdins,
> VoIP Project Manager / Developer,
> IQ Labs Inc,
>atis at iq-labs.net> Skype: atis.lezdins
> Cell Phone: +371 28806004
> Cell Phone: +1 800 7300689
> Work phone: +1 800 7502835
>>>> ------------------------------
>> Message: 20
> Date: Fri, 21 Nov 2008 13:46:00 -0200
> From: "Gonzalo Servat" <gservat at gmail.com>
> Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000
> extensions), preferably at universities
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <dcc007e10811210746s60e8d957i649106883a40ed3b at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>> On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller <noahisaacmiller at gmail.com>wrote:
>>> [..snip..]
>> With that many extensions, I'll second using a SIP registrar like
>> Freeswitch or OpenSer. Just use asterisk to provide the services.
>>>> Is Asterisk even needed?
>> - Gonzalo
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081121/e72dd610/attachment-0001.htm>> ------------------------------
>> Message: 21
> Date: Fri, 21 Nov 2008 09:46:27 -0600
> From: "Danny Nicholas" <danny at debsinc.com>
> Subject: Re: [asterisk-users] Limit the number of users in a
> meetmeconference?
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <3D33EB687590414696C75D9209ACF1F1 at db0005>
> Content-Type: text/plain; charset="us-ascii"
>> Armed with a little more information, here is a more realistic reply.
> In the 1.6.0.1 code, app_meetme.c defines maxusers in line 369 and sets the
> max value in line 870 to 0x7fffffff.
> Therefore changing line 870 would allow you to limit the maxusers.
>> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Noah Miller
> Sent: Friday, November 21, 2008 9:05 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Limit the number of users in a
> meetmeconference?
>> Hi Dan -
>>>> I found the "maxusers" defined in meetme.c, but I'm
>>> not sure how this value is set. Does anybody know
>>> if one can limit the number of users permitted in a
>>> meetme conference? I know there's MeetmeCount(), but
>>> I'd rather avoid the dialplan logic and just set
>>> maxusers instead.
>>>> That feature is primarily used with RealTime conferences.
>> The maxusers value is read from a database and enforced
>> on RealTime enable conferences. This presumes you are
>> looking at 1.6.X or Trunk code...
>> Ah. No realtime for me, so I guess I'll just stick with using
> MeetmeCount() in the dialplan. Thanks for the info!
>>> - Noah
>> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users>>>>> ------------------------------
>> Message: 22
> Date: Fri, 21 Nov 2008 10:48:57 -0500
> From: Alex Balashov <abalashov at evaristesys.com>
> Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000
> extensions), preferably at universities
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4926D869.2080305 at evaristesys.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>> Gonzalo Servat wrote:
>> On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller <noahisaacmiller at gmail.com>> <mailto:noahisaacmiller at gmail.com>> wrote:
>>>> [..snip..]
>>>> With that many extensions, I'll second using a SIP registrar like
>> Freeswitch or OpenSer. Just use asterisk to provide the services.
>>>>>> Is Asterisk even needed?
>> Potentially, no. But if you intend to provide subscriber/PBX features,
> it is needed as a UA feature box(s).
>> --
> Alex Balashov
> Evariste Systems
> Web : http://www.evaristesys.com/> Tel : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>>>> ------------------------------
>> Message: 23
> Date: Fri, 21 Nov 2008 11:14:57 -0500
> From: RE Kushner List Account <lists at darl.com>
> Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000
> extensions), preferably at universities
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4926DE81.50206 at darl.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>> Noah Miller wrote:
>>>>> and is only one of the roads that
>>> leads to Hell (I prefer Patterson Lake Road myself since I drive in from
>>> the North East).
>>>>>>> Hmm. You must live near Ann Arbor.
>>>>>> No, northern suburbs of Detroit. M-59 to US-23 S to M-36 W..To S.
> Howell St..Patterson Lake Rd..To Hell....
>> Ann Arbor is quite a bit South of Hell. Actually it's been some time
> since I've been to Hell but I'm sure it's frozen over today ;-)
>> -Ron
>>>>> ------------------------------
>> Message: 24
> Date: Fri, 21 Nov 2008 11:28:18 -0500
> From: Jerry Geis <geisj at pagestation.com>
> Subject: [asterisk-users] upgrade from 1.2 to 1.4 and now half channel
> audio
> To: asterisk-users at lists.digium.com> Message-ID: <4926E1A2.1000001 at pagestation.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>> Hi all,
>> I upgraded from asterisk 1.2.23 and zaptel 1.2.19
> to asterisk 1.4.18 and zaptel 1.4.12.1
> I use polycom 501 phones internally.
>> Everything seems fine. I can pick up the phone and call out,
> calls coming in work just fine.
>> The issue I see is when the system first calls me,
> then calls someone else. This works if its polycom to polycom. I hear
> audio full channel.
> If I do polycom to external line like a cell I only get HALF channel audio.
> At this time they can hear me but I cannot hear them.
>> What might this be???
>> Jerry
>>>> ------------------------------
>> Message: 25
> Date: Fri, 21 Nov 2008 17:32:22 +0100
> From: Olivier <oza-4h07 at myamail.com>
> Subject: [asterisk-users] OT - SIP message encoding to enhance text
> display
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <442fbb120811210832h13e5b054ncf57a66c8a5dcb47 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>> Hi,
>> I've read RFC3428 which presents SIP MESSAGE.
> Is there any extension or encoding scheme working with SIP MESSAGE that
> would enhance text display with blinking or underlining attributes ?
> This could be useful to notify SIP hardphone users with some important
> events such being in Do Not Disturb mode.
>> Regards
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